The number of applications is increasing which transmit video data and audio data as IP (internet protocol) packets from a transmitting side to a receiving side, and decompose and reproduce the video data and audio data at the receiving side. For example, IP telephone (VoIP) on the Internet, and video delivery of teleconference, movie and live may be enumerated. In these applications, in order for maintaining the quality of video or audio, it is an important factor to suppress occurring of jitter and packet loss. The “jitter” means fluctuation of the time length need for receiving, at the receiving side, the packet transmitted from the transmitting side.
If a large jitter occurs, the receiving interval of the video data or audio data is enlarged, and thus a quality degradation is encountered wherein a normal reproduction is impossible. The packet loss is a phenomenon such that the transmitted packet disappears due to a network congestion etc., and thus is not received at the receiving side. For this phenomenon, an application on the receiving side cannot receive the packets needed for reproduction, which causes a trouble in the reproduction.
As a communication protocol in the application such as for the video and audio delivery, a protocol without a retransmission control is mostly used. UDP (user datagram packet) may be exemplified as a communication protocol generally used. The UDP is a communication protocol that is used for achieving a weight saving and a high speed. More specifically, if a transmitted packet is lost before reaching a terminal on the receiving side, the lost packet is not retransmitted. For this reason, although there may be some data that are not replicated at the receiving side, a delay accompanied by the retransmission does not occur. However, in consideration that there may be a quality degradation due to the absence of the retransmission, a method has been proposed and realized for improving the tolerance to the packet loss.
Patent Publication JP-2002-9883A proposes a method for preventing the quality degradation caused by the packet loss by using an automatic repeat request (ARQ) scheme, and a forward error correction (FEC) scheme.
The ARQ scheme is such that if a packet loss is detected at the receiving side, the receiving side requests retransmission of the corresponding packet from the transmitting side to thereby recover the packet loss. The FEC scheme is such that the transmitting side prepares a redundant packet or packets beforehand and transmits the same together with the original data to the receiving side. If a packet loss occurs, it is possible to use the redundant packet or packets to recover the packet loss at the receiving side.
Conventionally, one of the ARQ scheme and FEC scheme had to be fixedly selected as a means to recover the packet loss. JP-2002-9883A proposes an appropriate changeover between the ARQ scheme and the FEC scheme based on the state of the network. In JP-2002-9883A, an error rate (number of error bits/total number of transmitted bits) is used as the reference value for the changeover between the ARQ scheme and the FEC scheme. This is based on the following reasons.
The ARQ scheme functions effectively if the total number of transmitted bits is larger and the number of error bits is smaller; however, the ARQ scheme has a poor efficiency if the total number of transmitted bits is smaller and the number of error bits is larger. On the other hand, the FEC scheme functions effectively if the total number of transmitted bits is smaller and the number of error bits is larger; however, the ARQ scheme has a poor efficiency if the total number of bits is larger and the number of error bits is smaller, due to a higher overhead of the FEC packets.
For that reason, JP-2002-9883A recites that setting of a threshold based on the error rate allows the changeover to be performed between the ARQ scheme and the FEC scheme based on whether or not the error rate exceeds the threshold.
There is another index based on which the network is to be evaluated, other than the error rate. For example, as described before, there is jitter as a parameter that affects the quality of the video and audio delivery. The problem in the above patent publication will be conspicuous under the circumstance of a larger jitter and a higher packet loss rate.
The technique according to JP-2002-9883A will employ the FEC scheme based on a higher degree of the packet loss rate. It is necessary in the FEC scheme to await arrival of a redundant packet for performing recovery of the packet loss. However, the larger jitter may delay the arrival of the redundant packet. Thus, even though recovery of the lost packet itself may be achieved, there may be a case where the reproduction cannot be performed in time. Accordingly, it cannot be judged in this case that the FEC scheme is the optimum. In other words, the error rate alone is insufficient as the index for the judgment of the changeover between the ARQ scheme and the FEC scheme, and accordingly, there may arise a quality degradation during the reproduction in the technique of JP-2002-9883A.